Advanced SIP settings

Display Name this will be used as your Caller ID if supported by your VoIP provider

Auth Username some providers require you to enter a separate authorization username. Enter it here

Dialling Rules please see the written tutorial on Dialling rules for more details

DTMF Mode there are three settings RFC2833 Inband DTMF (audio) and SIP INFO. DTMF is sent when you press a digit on the in-call keypad when dealing with an auto-attendant ( such as when you dial your bank and are asked to choose from a set of options).

If you find DTMF tones are not being recognized changing to one of the other methods should resolve the problem.

WiFi Codecs the G711u G711a GSM and iLBC codecs are supported. Codecs at the top of the list will be offered to the remote end as the preferred codec. The softphone will always choose the most preferred codec offered by the remote end. Tap the 'Move' button in the top right corner, this will allow you to tap and drag the rows. To disable a codec move it to the disabled section. Either the G711u or G711a will give the best audio quality but if the network connection is poor it is best to use one of the lower bit rate codecs such as GSM or iLBC.

If codec negotiation fails you will receive a 488 error and the call will fail to be established. To avoid this try to have as many codecs enabled as possible.

3G codecs the points mentioned above in the WiFi codecs section apply equally here.

The important thing to bear in mind is that typically G711u and G711a have too high bandwidth to be used reliably on 3G networks. if you are finding that audio quality is poor over 3G, check which codec is being negotiated. The codec being used during the call is displayed in the top right corner of the in-call display just below the account name. If the codec is G711u or G711a it would be best to completely disable these for 3G calls. Simply drag both to the disabled section as shown in the image. You should then have good quality calls over 3G using either GSM or iLBC. Please check whether your provider supports either of these lower bit rate codecs.

We plan to offer G729 as an in-app purchase as soon as possible as this will also offer good quality over 3G networks.

Transport the app currently supports UDP and TCP transport. UDP the default setting is used by most VoIP providers. We plan to support TLS in a future update.

Incoming Calls if set to 'On' (which is the default setting) the app will register for incoming calls and receive calls whilst the app is in the foreground or background ( background will work so long as 'Keep Awake Mode' is set to Always in the preferences).

If set to 'Off' no register attempts will be made. Only use this if you are only going to make outgoing calls on this account. This will save battery life as no SIP REGISTER messages or keepalives are sent.

'On with Push' enables use of Push Notifications for the account. This option is grayed out if the 'Keep Awake Mode' setting is set to 'Always'. in this mode there is no need for Push support. When the app is moved to the background the user's credentials are transferred over a secure HTTPS connection to our server. Our server then registers and listens for incoming calls to the account and notifies the device when it receives a call. Please note Push notifications need to be enabled in the device Settings application. Missed calls are notified via Push as well and will appear in the call log.

Global IP if enabled the app will present its public IP address in the SIP signalling messages to your provider's server. This allows the provider to route the messages back to the app. However using the public address may not work in all cases. For example some providers' servers require you to register with a private IP address. Turning the switch to 'Off' will cause the app to use a private IP. This setting applies when using the app over a WiFi network.

Global IP 3G same as above except this setting applies when using the app over the cellular network

Voicemail Number if your provider supports Voicemail, entering the number here allows you to dial your voicemail directly from the dialler screen by tapping the VM (Voicemail) button in the bottom left corner.

Proxy Address and Port if your provider requires use of a proxy enter it here otherwise leave empty.

Forward To Call forwarding needs to be enabled in the global preferences. Overrides the number in the preferences settings so incoming calls to this SIP account will be forwarded to this number.

UDP Keepalive Enables sending of UDP keepalive packets. These are used to keep the pinhole open on your NAT/Firewall router so you receive incoming calls reliably.

Keepalive Interval this interval specifies how often the UDP keepalive messages are sent. A higher value will save battery life however if you find you are missing incoming calls you should lower the value. The default value of 15 seconds should be low enough to work with most NAT devices.

RTP Port Start/End This defines the range of ports used for RTP sockets.

Expiry Period this is interval in seconds that the application will attempt to REGISTER with your provider. Defaults to 600 seconds. The server may change this value however.

STUN Enabled if enabled the app will use the STUN server specified to discover its public IP address. This public IP is sent to your provider's server to inform it where to route RTP audio packets. If disabled a private IP address is used. Enabling STUN can often be used to resolve one way audio issues. This setting only applies when connected on WiFi Networks.

STUN 3G Enabled Same as above except only applies on 3G networks

Outbound Proxy see below

SIP Port By default the app selects a random port for sending SIP signalling messages to your provider. Specify a port here to force the app to listen to SIP messages on a particular port. Only applies when using the UDP transport.

Outbound Proxy In most cases you should not have to enter anything here. If your provider requires use of a proxy please use the Proxy Address field described earlier.

Use this option if you need to route all SIP messages through a specific server.